The present invention generally relates to wireless communication. More specifically, the invention relates to a method and apparatus for multiplexing real-time users in a packet switched radio communication system.
There is presently ongoing a paradigm shift in telecommunication. Historically, the telecommunications industry has been focusing on voice communication over fixed lines or radio communication links like, e.g., cellular telephony systems like Global System for Mobile communication (GSM). Communication has typically been transmitted in a circuit switched manner, i.e., with dedicated connections between users or end nodes. Circuit switched communication requires continuous allocation of physical transmission resources, or communication channels, for the whole duration of a connection, regardless of the actual use of the connection.
With the explosive growth of Internet traffic however, the focus has shifted towards more efficient ways of transferring data communication in a telecommunication network. Packet switched communication protocols has been developed, e.g., General Packet Radio Service (GPRS) to be used together with GSM and the Time Division Multiple Access (TDMA) system compliant to the TIA/EIA-136 standard. The advantage with these packet switched communication protocols is that there is no need to have physical transmission resources reserved for users that are not making use of it. For example, a user may share a transmission resource with one or several other users and occupy the resource only when there is user data to send. If there is no data to send during certain periods, other users may utilize the transmission resources. This is a more efficient way of allocating users onto physical channels than the circuit switched strategy, where a user is always a sole owner of a communication channel.
With the identification of packet switched methods as being an efficient way of transferring data, the next step is basically a step back. The focus is again on voice, but it is also a step forward in that the aim is now set on voice over packet switched communication, or more generally, real-time services over packet switched communication channels. With this and other aims, there will be a large variety of services carried over packet switched communication channels, services with completely different demands in terms of delay, delay variations (jitter) and error rates. For example, a web browsing session may not suffer seriously from being slightly delayed, it is however important that the transfer is error free. For voice communication, it is basically the other way around; a voice conversation is extremely sensitive to delay and delay variations but may perhaps tolerate a non-zero error rate and still provide acceptable quality.
In the Universal Mobile Telecommunications System (UMTS), there are four proposed classes defined to further characterize different services and the respective Quality of Service (QoS) demands: conversational, streaming, interactive and background. One main distinguishing factor between these classes is delay related. The conversational class is intended for delay sensitive traffic, such as speech, while the background class is the most delay insensitive class. Conversational and streaming classes are intended to be used to carry real-time traffic flows and interactive and background classes are intended to carry, e.g., Internet applications like WWW-browsing, file transfer and e-mail services.
As voice communication involves constraints on delay, it does not tolerate the sharing of a transmission resource, or physical channel, as liberally as the fundamentals of packet switched communication allow. It is necessary to introduce priority for voice users over, e.g., a background user on the same channel, such that the real-time aspects of the voice connection may be maintained.
In an exemplar voice call, there are typically periods of silence in one direction when the other direction speaks, and vice versa. With circuit switched radio communication connections, it is possible to utilize these silent periods and decrease the output power from the transmitter while a voice stream from a speech coder is paused. This will mean a system gain in terms of less interference. The physical communication channels, e.g., in terms of frequency, timeslot or code is however still occupied. There may however be even more to gain if other users could be multiplexed onto the same physical channel during these speech pauses. By using packet switched methods for transferring voice communication of the conversational class, it will become possible to more efficiently make use of the transmission resources while in a period of speech silence. One way to do this is to allocate the resources to a best effort user, e.g., of the background or interactive class, while in a silence period and maintain the high priority for the conversational class user. Thus, it will be easy to, as soon as a silence period is interrupted by a speech period, prioritize allocation of the conversational class again. With this flexible method of allocating shared resources, it will be possible to allocate more users than the number of available transmission resources or channels. If there is a high number of transmission resources, it may even be possible to allocate more voice users than the number of channels, assuming that it is highly unlikely that all users need transmission resources at the same time. This strategy is usually referred to as statistical multiplexing.
For the Adaptive Multi-Rate (AMR) speech coder structure of GSM, as in many other speech coders designed for circuit switched connections, the silent periods discussed above are not completely transmission free, i.e., the transmission resources are still utilized. During a silent period, when no speech is processed, the speech coder generates what can be referred to as a Silence Descriptor (SID). This silence descriptor is transmitted according to some repetition rate in order to generate “comfort noise” in the receiving end. It is typically the case in a voice communication that there is no complete silence, and to “simulate” the noise usually present in the surroundings of the silent speaker, SIDs are transmitted with a certain repetition rate. The SIDs defined for circuit switched speech are traditionally transmitted on the same physical resource as the regular voice communication.
If a packet switched system is considered, the silent periods should optimally enable allocation of other users onto the physical communication channel, e.g., background or interactive class users. It would of course be possible to do this and still transmit SIDs from a conversational user also. However, if one consider utilizing a communication channel for more than one conversational user in one way or another (e.g., statistical multiplexing), the SID transmissions that are continuously repeated with some repetition pattern will pose a problem, since a continuous allocation for e.g., another real-time user will be impossible. There is thus a need to develop and prepare techniques to more efficiently allocate resources and allow a more flexible scheduling, than what is possible with the presently used SID techniques.